OLD | NEW |
1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2004 Google Inc. | 3 * Copyright 2004 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
(...skipping 120 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
131 | 131 |
132 VoEWrapper* voe() { return voe_wrapper_.get(); } | 132 VoEWrapper* voe() { return voe_wrapper_.get(); } |
133 int GetLastEngineError(); | 133 int GetLastEngineError(); |
134 | 134 |
135 // Set the external ADM. This can only be called before Init. | 135 // Set the external ADM. This can only be called before Init. |
136 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); | 136 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); |
137 | 137 |
138 // Starts AEC dump using existing file. | 138 // Starts AEC dump using existing file. |
139 bool StartAecDump(rtc::PlatformFile file); | 139 bool StartAecDump(rtc::PlatformFile file); |
140 | 140 |
141 // Check whether the supplied trace should be ignored. | |
142 bool ShouldIgnoreTrace(const std::string& trace); | |
143 | |
144 // Create a VoiceEngine Channel. | 141 // Create a VoiceEngine Channel. |
145 int CreateMediaVoiceChannel(); | 142 int CreateMediaVoiceChannel(); |
146 | 143 |
147 private: | 144 private: |
148 typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList; | 145 typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList; |
149 typedef sigslot:: | 146 typedef sigslot:: |
150 signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal; | 147 signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal; |
151 | 148 |
152 void Construct(); | 149 void Construct(); |
153 void ConstructCodecs(); | 150 void ConstructCodecs(); |
(...skipping 282 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
436 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 433 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
437 | 434 |
438 // Do not lock this on the VoE media processor thread; potential for deadlock | 435 // Do not lock this on the VoE media processor thread; potential for deadlock |
439 // exists. | 436 // exists. |
440 mutable rtc::CriticalSection receive_channels_cs_; | 437 mutable rtc::CriticalSection receive_channels_cs_; |
441 }; | 438 }; |
442 | 439 |
443 } // namespace cricket | 440 } // namespace cricket |
444 | 441 |
445 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 442 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
OLD | NEW |